Support for Signal calls.

Merge in RedPhone

// FREEBIE

jni/webrtc/modules/audio_coding/neteq/tools/packet_source.h

@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
+
+#include <bitset>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+class Packet;
+
+// Interface class for an object delivering RTP packets to test applications.
+class PacketSource {
+ public:
+  PacketSource() {}
+  virtual ~PacketSource() {}
+
+  // Returns a pointer to the next packet. Returns NULL if the source is
+  // depleted, or if an error occurred.
+  virtual Packet* NextPacket() = 0;
+
+  virtual void FilterOutPayloadType(uint8_t payload_type) {
+    filter_.set(payload_type, true);
+  }
+
+ protected:
+  std::bitset<128> filter_;  // Payload type is 7 bits in the RFC.
+
+ private:
+  DISALLOW_COPY_AND_ASSIGN(PacketSource);
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_